1. Field of the Invention
The present invention relates to an apparatus for forming musical tone signals to be used as a sound source for electronic musical instruments and the like.
2. Description of the Prior Art
In electronic musical instruments and the like, to form tone signals approximated to those by natural musical instruments there have been proposed physical model sound sources that simulate the sound generation principle of natural musical instruments.
These physical model sound sources, for example, one which simulates some woodwind instrument would be so arranged that it simulates the mouthpiece part that generates aerial vibrations by a non-linear circuit while the tone sound source simulates the pipe part, which allows aerial vibrations to propagate therethrough and is resonant only with aerial vibrations at certain frequencies, by a linear circuit.
FIGS. 8 and 9 are views showing a common physical model sound source that simulates the sound generation principle of woodwind instruments.
FIG. 8 is a view showing the construction of its non-linear part. This non-linear part simulates of the principle of forming aerial vibrations in single reed instruments such as saxophones. The non-linear part MP has a subtracter A4, which outputs a signal corresponding to a differential-pressure for causing a displacement of the reed by subtracting a breath pressure signal PRES, which is inputted from the mouthpiece of a playing control manipulator, from a feedback waveform signal FR, which has been fed back from the linear part. This signal corresponding to differential-pressure (differential-pressure signal) is inputted into a low-pass filter L and a non-linear table T2. In the non-linear table T2, input/output characteristic data is stored, such as shown in FIG. 8(c), which simulates the fact that even if the differential pressure becomes larger, the flow velocity saturates in such a narrow air path that the differential pressure and the flow velocity are not in proportion to each other. The low-pass filter L produces an output in which the high-band component of the differential-pressure signal has been removed. This is because the reed of the woodwind instrument will not respond to the high frequency component of input signal. The output of the low-pass filter L is inputted into an adder A3. The adder A3 receives an input of an embouchure signal EMBS, which represents how the mouth is tightened, i.e., pressure applied to the mouthpiece. The value resulting from adding these inputs is inputted into a non-linear table T1. The non-linear table T1, which simulates the amount of displacement of the reed with respect to an applied pressure, stores data shown in FIG. 8 (B). The output of the non-linear table T1 is a signal representing the area of air path at the reed tip of the mouthpiece. The output of the non-linear table T1 is inputted into a multiplier M3, to which the output of the non-linear table T2 (i.e. corrected differential pressure) is also inputted. With this arrangement, the multiplier M3 multiplies the differential pressure and the path area together, thereby calculating the flow velocity of air. The resulting output is further inputted into a multiplier M4, which produces an output by multiplying data that represents the flow velocity of the aforementioned air with a factor k that represents the impedance (air resistance) within the mouthpiece. The produced data is outputted to the pipe part as a tone pressure signal (traveling wave) FD. The circuitry described above allows a simulation of the process through which the flow velocity of air varies periodically to form compression waves.
FIG. 9 is a view showing the construction of the linear circuit. This linear circuit simulates the resonant state of air columns in the pipe body of a woodwind instrument (column-shaped air groups present in the pipe). The circuit is made up of a plurality of tone holes THn, pipe parts Dn which interconnect these tone holes, and a pipe tip TRM. It is noted that in this figure, only one piece of tone hole (TH1) is shown and two pipe parts (D1 and D2) are shown. These circuits are connected in series. The pipe parts Dn simulate a part of the pipe body with delay circuits DFn and DRn. More specifically, time required for sound waves (compression waves) to propagate increases depending on the length of the pipe, so that the delay time of the delay circuits corresponds to the length of the intervals of the tone holes. DF in these delay circuits represents a delay circuit for transmission of traveling wave signals and DR represents a delay circuit for transmission of reflected wave signals. TH simulates scattering of pressure waves, or forced formation of nodes of air vibration in the vicinity of tone holes. M1 and M2 represent multipliers; A1 and A2 represents subtracters; Aj represents an adder. In the TRM, a low-pass filter ML simulates attenuation in high band involved in reflection of aerial vibrations, while an inverter IV simulates the 180- degree phase inversion when reflection occurs at the open end.
Factors a1 and a2 of the THn, which are inputted into the multipliers M1 and M2 and then multiplied to the traveling wave signal and the reflected wave signal, respectively, are given different values depending on whether the tone hole is open or closed, which allows a simulation of change in tone pitch due to finger operation with the pipe instrument. When the tone hole is open, the following parameters will be given: EQU a1=2.PHI..sub.1.sup.2 /(.PHI..sub.1.sup.2 +.PHI..sub.2.sup.2 +.PHI..sub.3.sup.2) EQU a2=2.PHI..sub.2.sup.2 /(.PHI..sub.1.sup.2 +.PHI..sub.2.sup.2 +.PHI..sub.3.sup.2)
Meanwhile, when the tone hole is closed, the parameters will be given as: EQU a1=2.PHI..sub.1.sup.2 /(.PHI..sub.1.sup.2 +.PHI..sub.2.sup.2) EQU a2=2.PHI..sub.2.sup.2 /(.PHI..sub.1.sup.2 +.PHI..sub.2.sup.2)
where .PHI.1, .PHI.2, and .PHI.3 represent a front diameter of the resonant pipe, a rear diameter of the resonant pipe, and a diameter of the tone hole, respectively. Determination manner of the parameters is described in detail in U.S. patent application Ser. No. 07/511,060 filed on Apr.19, 1990.
Alternatively, when any string instrument is simulated, the friction of strings by the bows is simulated by a non-linear circuit, while vibration of strings is simulated by a linear circuit.
As shown above, the above-mentioned physical model sound sources have been provided in such a construction that all the parts for forming tone signals from the non-linear to linear circuits are in an integral form or a physically fixed form. In the case of woodwind instruments, the principle of forming aerial vibrations by the mouthpiece part would be approximately similar among single reed instruments such as a plurality of types of saxophones and clarinets, yet the length of pipe and the transmission characteristic of aerial vibrations are slightly different.
Likewise, when a string instrument is simulated, the vibration of strings is based on similar principle among plural kinds of string instruments, that is, friction by bows, the ways of resounding are different because the length and thickness of strings as well as in the size of the body are different among those instruments. These factors can be approximately simulated by changing the parameters of the delay circuits or others.
However, the conventional physical model sound sources, are constructed wholly as an integral unit as described above, can allow one sound source to generate sounds in only one type of instrument. Therefore, forming tones (tone signals) of a plurality of types of instruments necessitates providing plural sets of the entire circuit for respective types of instruments.
As another aspect of common electronic musical instruments, there have been adopted sound source circuits in which tone signals are formed in only one method. As a result of this, the CPU has only to drive a fixed tone formation algorithm, permitting use of fixed programs including a control program for tone generation and a program for supporting tone color edition. Thus, it has been taken for granted that such data is stored in a ROM within the instrument body.
Meanwhile, although even a sound source circuit of the FM synthesizing method would be able to generate various types of tones (waveforms), these tones could be realized by changing parameters for synthesizing (voicing data) in various ways while the method of synthesizing would not be changed. As a result, even in an electronic musical instrument which allows any external memory such as a cartridge to be connected thereto, only voicing data, which does not change any tone synthesizing method, can be fed from the external memory. Also, there have been proposed electronic musical instruments in which a plurality of control programs are stored in a memory and any one of the program is selected to change the tone formation algorithm. Yet this method would allow the selection only in a pre-stored range.
To form tone signals having their own characteristics, it is desirable that the tone formation algorithm itself be changed for each tone, which was impossible for conventional electronic musical instruments.